How a Phone System Integration Works

A phone system integration depends on the following components to be successful:

Lines and cables necessary to make physical connections (for PIMG/TIMG integrations) or a network connection (in Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, SIP proxy servers, and QSIG-enabled phone systems). Depending on the type of integration, the phone system connects through different combinations of lines. See the applicable section for more information:

Digital Integration with Digital PIMG Units

The phone system sends call information, MWI requests, and voice connections through the digital lines, which connect the phone system to the PIMG units (media gateways). The PIMG units communicate with the Cisco Unity Connection server through the LAN or WAN by using Session Initiation Protocol (SIP). Figure 10-2 shows the connections used in a digital integration by using digital PIMG units.

Figure 10-2 Connections for a Digital Integration by Using Digital PIMG Units

DTMF Integration with Analog PIMG Units

The phone system sends call information, MWI requests, and voice connections through the analog lines, which connect the phone system to the PIMG units (media gateways). The PIMG units communicate with the Cisco Unity Connection server through the LAN or WAN by using Session Initiation Protocol (SIP). Figure 10-3 shows the connections for a DTMF integration by using analog PIMG units.

Figure 10-3 Connections for a DTMF Integration by Using Analog PIMG Units

Serial (SMDI, MCI, or MD-110) Integration with Analog PIMG Units

The phone system sends call information and MWI requests through the data link, which is an RS-232 serial cable that connects the phone system and the master PIMG unit (media gateways). Voice connections are sent through the analog lines between the phone system and the PIMG units. The PIMG units communicate with the Cisco Unity Connection server through the LAN or WAN by using Session Initialization Protocol (SIP). Figure 10-4 shows the connections for a serial integration by using analog PIMG units.

Figure 10-4 Connections for a Serial (SMDI, MCI, or MD-110) Integration by Using Analog PIMG Units

Note When you use multiple PIMG units, one PIMG unit must be designated the master PIMG unit, which is connected to the serial cable from the phone system. It is not possible to “daisy chain” the serial ports on the PIMG units.

You can add a secondary master PIMG unit to an integration. For details, see the “Appendix: Adding a Secondary Master PIMG Unit” appendix of the PIMG Integration Guide for Cisco Unity Connection Release 8.x, at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/integration/guide/pimg/cucintpimg.html.

TIMG Serial (SMDI, MCI, or MD-110) Integration

The TIMG integration uses one or more TIMG units between circuit-switched phone systems and IP networks. On the circuit-switched phone system side, there is a T1-CAS interface. On the IP side, there is a SIP interface, which is how Cisco Unity Connection communicates with the TIMG units. To Connection, the integration is essentially a SIP integration. Connection communicates with the TIMG units over the IP network by using SIP and RTP protocols. The TIMG units communicate with the circuit-switched phone system over the phone network by using serial protocols (SMDI, MCI, or MD-110).

The phone system sends call information and MWI requests through the data link, which is an RS-232 serial cable that connects the phone system and the master TIMG unit. Voice connections are sent through the T1 digital lines between the phone system and the TIMG units. The TIMG units communicate with the Cisco Unity Connection server through the LAN or WAN by using Session Initialization Protocol (SIP). Figure 10-5 shows the connections for a serial integration by using TIMG units.

Figure 10-5 Connections for a Serial Integration by Using TIMG Units

TIMG In-Band Integration

The phone system sends call information, MWI requests, and voice connections through the T1 digital lines, which connect the phone system and the TIMG units. The TIMG units communicate with the Cisco Unity Connection server through the LAN or WAN by using Session Initialization Protocol (SIP). Figure 10-6 shows the required connections for an in-band integration by using TIMG units.

Figure 10-6 Connections for an In-Band Integration by Using TIMG Units

Settings in the Phone System and in Cisco Unity Connection

For an integration to be successful, Cisco Unity Connection and the phone system must know the connections to use (for example, IP addresses and channels) and the expected method of communication (for example, IP packets, serial packets, and DTMF tones). Certain integrations require specific codes or extensions for turning MWIs on and off.

Call Information Exchanged by the Phone System and Cisco Unity Connection

The phone system and Cisco Unity Connection exchange call information to manage calls and to make the integration features possible. With each call, the following call information is typically passed between the phone system and Connection:

The extension of the called party.

The extension of the calling party (for internal calls) or the phone number of the calling party (if it is an external call and the phone system supports caller ID).

The reason for the forward (the extension is busy, does not answer, or is set to forward all calls). There is also a reason code for Direct Calls.

Cisco Unified Communications Manager SCCP and SIP trunk integrations can also provide the following call information:

Called number

First redirecting number

Last redirecting number

Note Connection can use either the first redirecting number or last redirecting number, depending on the setting of the Use Last (Rather than First) Redirecting Number for Routing Incoming Call check box on the System Settings > Advanced > Conversations page in Cisco Unity Connection Administration.

If the phone system sends the necessary information and if Connection is configured correctly, an integration can provide the following integration functionality:

Call forward to personal greeting

Call forward to busy greeting

Caller ID

Easy message access (a user can retrieve messages without entering an ID because Connection identifies the user based on the extension from which the call originated; a password may be required)

Identified user messaging (Connection identifies the user who leaves a message during a forwarded internal call, based on the extension from which the call originated)

Message waiting indication (MWI)

Call Control

The phone system uses a set of signals to set up, monitor, and release connections for a call. Cisco Unity Connection monitors call control signals to determine the state of the call, and uses these signals to respond appropriately to phone system actions and to communicate with the phone system. For example, a caller who is recording a message might hang up, so Connection detects that the call has ended and stops recording.

Depending on the phone system, the following types of call control signals are used:

For SIP trunk integrations, Cisco Unified CM sends SIP messages, and Connection sends SIP responses when a call is set up or terminated.

Circuit-switched phone system through PIMG/TIMG units

The phone system sends messages to the PIMG or TIMG units (media gateways), which send the applicable SIP messages to Connection. Connection sends SIP responses when a call is set up or terminated, and the PIMG or TIMG units communicate with the phone system.

Note If Extension A calls Extension B and if the Display Name or ASCII Alerting Name of Extension B contains the word "Vociemail", then the call arrives on Cisco Unity Connection as a Direct Call rather than as a forwarded Call.

Sample Path for a Call from the Phone System to a User

The following steps give an overview of a sample path that an external call can take when traveling from the phone system to a user.

1. For Cisco Unified Communications Manager, when an external call arrives, the gateway sends the call over the LAN or WAN to Cisco Unified CM. Cisco Unified CM routes the call to the Cisco Unity Connection voice mail pilot number.

For circuit-switched phone systems, when an external call arrives via the PSTN, TI/PRI, DID or LS/GS analog trunks, the phone system routes the call to the Cisco Unity Connection voice mail pilot number.

2. The phone system routes the call to an available Cisco Unity Connection voice messaging port.

3. Connection answers the call and plays the opening greeting.

4. During the opening greeting, the caller enters an extension. For example, the caller enters 1234 to reach a person at that extension.

5. Connection notifies the phone system that there is a call for extension 1234.

6. Depending on whether Connection is set up to perform a release transfer or a supervised transfer, the following occurs:

Release transfer (blind transfer)

Connection passes the call to the phone system, and the phone system sends the call to extension 1234 without waiting to determine whether the line is available. Then the phone system and Connection drop out of the loop. In this configuration, if the customer wants Connection to take a message when a line is busy or unanswered, each phone must be configured to forward calls to Connection when the line is busy or unanswered.

Supervised transfer

While Connection holds the call, the phone system attempts to establish a connection with extension 1234.

If the line is available, the phone system connects the call from Connection to extension 1234. The phone system and Connection drop out of the loop, and the call is connected directly from the original caller to extension 1234.

If the line is busy or unanswered, the phone system gives that information to Connection, and Connection performs the operation the user has specified. For example, Connection takes a message.

Cisco Unity Connection and Cisco Unified Communications Manager deployment models, including single-site messaging, centralized messaging, and distributed messaging, can be combined to suit customer requirements. When choosing a deployment model, you must consider a range of issues, for example:

Centralized messaging allows you to consolidate servers and administration, but requires that you plan for access to voice messages in the event of WAN outages and that you perform the appropriate QOS/capacity planning for voice-messaging traffic and call traffic.

A potential point of vulnerability for a Cisco Unity Connection system is the connection between Connection and Cisco Unified Communications Manager. Possible threats include:

Man-in-the-middle attacks, in which an attacker intercepts and changes the data flowing between Cisco Unified CM and Connection voice messaging ports.

Network traffic sniffing, in which an attacker captures phone conversations and signaling information that flow between Cisco Unified CM, the Connection voice messaging ports, and IP phones that are managed by Cisco Unified CM.

Cisco Unified Communications Manager Security Features

Cisco Unified Communications Manager Release 4.1(3) or later for SCCP integrations or Cisco Unified Communications Manager Release 5.x or later for SIP trunk integrations can secure the connection with Cisco Unity Connection against security threats. The Cisco Unified CM security features that Connection can take advantage of are described in
Table 10-2.

Table 10-2 Cisco Unified Communications Manager Security Features That Are Used by Cisco Unity Connection

Security Feature

Description

Signaling authentication

Uses the Transport Layer Security (TLS) protocol to validate that no tampering has occurred to signaling packets during transmission. Signaling authentication relies on the creation of the Cisco Certificate Trust List (CTL) file.

Validates the identity of the device. This process occurs between Cisco Unified CM and the Connection voice messaging ports when each device accepts the certificate of the other device. When the certificates are accepted, a secure connection between the devices is established. Device authentication relies on the creation of the Cisco Certificate Trust List (CTL) file.

Uses cryptographic methods to protect (through encryption) the confidentiality of all SCCP and SIP signaling messages that are sent between the Connection voice messaging ports and Cisco Unified CM. Signaling encryption ensures that the information that pertains to the parties, DTMF digits that are entered by the parties, call status, media encryption keys, and so on are protected against unintended or unauthorized access.

Uses Secure Real Time Protocol (SRTP) as defined in IETF RFC 3711 to ensure that only the intended recipient can interpret the media streams between the Connection voice messaging ports and endpoints (for example, phones or gateways). Only audio streams are encrypted. Media encryption creates a media master key pair for the devices, delivers the keys to Connection and the endpoint, and secures the delivery of the keys while the keys are in transport. Connection and the endpoint use the keys to encrypt and decrypt the media stream.

This feature protects against:

Man-in-the-middle attacks that listen to the media stream between Cisco Unified CM and the Connection voice messaging ports.

Network traffic sniffing that eavesdrops on phone conversations that flow between Cisco Unified CM, the Connection voice messaging ports, and IP phones that are managed by Cisco Unified CM.

Authentication and signaling encryption are required for media encryption; that is, if the devices do not support authentication and signaling encryption, media encryption cannot occur.

Note that Cisco Unified CM authentication and encryption protects only calls to Connection. Messages that are recorded on Connection are not protected by Cisco Unified CM authentication and encryption but can be protected by the Connection secure messaging feature.

A Connection server root certificate for each Connection server that uses authentication and/or encryption. A root certificate is valid for 20 years from the time it was created.

Connection voice messaging port or port group device certificates that are rooted in the Connection server root certificate, and voice messaging ports or port groups that are present when registering with the Cisco Unified CM server.

The process of authentication and encryption of Connection voice messaging SCCP ports occurs as follows:

The process of authentication and encryption of Connection voice messaging SIP port groups occurs as follows:

1. Each Connection voice messaging port group connects to the TFTP server, via TFTP port 69, downloads the CTL file, and extracts the certificates for all Cisco Unified CM servers.

2. Each Connection voice messaging port group establishes a network connection to the Cisco Unified CM TLS port. By default, the TLS port is 2443, though the port number is configurable.

3. Each Connection voice messaging port group establishes a TLS connection to the Cisco Unified CM server, at which time the device certificate is verified and the voice messaging port group is authenticated.

4. Each Connection voice messaging port group registers with the Cisco Unified CM server, specifying whether the voice messaging port group will also use media encryption.

When Data Is Encrypted

When a call is made between Cisco Unity Connection and Cisco Unified CM, the call-signaling messages and the media stream are handled in the following manner:

If both endpoints are set for encrypted mode, the call-signaling messages and the media stream are encrypted.

If one endpoint is set for authenticated mode and the other endpoint is set for encrypted mode, the call-signaling messages are authenticated. But neither the call-signaling messages nor the media stream are encrypted.

If one endpoint is set for non-secure mode and the other endpoint is set for encrypted mode, neither the call-signaling messages nor the media stream are encrypted.

The Security Mode settings in Cisco Unity Connection Administration determine how the ports handle call-signaling messages and whether encryption of the media stream is possible.
Table 10-3 describes the effect of the Security Mode settings on the Telephony Integrations > Port > Port Basics page for each port in an SCCP integration.

The integrity and privacy of call-signaling messages will not be ensured because call-signaling messages are sent as clear (unencrypted) text and are connected to Cisco Unified CM through a non-authenticated port rather than an authenticated TLS port.

In addition, the media stream cannot be encrypted.

Authenticated

The integrity of call-signaling messages is ensured because they are connected to Cisco Unified CM through an authenticated TLS port. However, the privacy of call-signaling messages is not ensured because they are sent as clear (unencrypted) text.

In addition, the media stream is not encrypted.

Encrypted

The integrity and privacy of call-signaling messages is ensured because they are connected to Cisco Unified CM through an authenticated TLS port, and the call-signaling messages are encrypted.

In addition, the media stream can be encrypted.

Caution Both endpoints must be registered in encrypted mode for the media stream to be encrypted. However, when one endpoint is set for non-secure or authenticated mode and the other endpoint is set for encrypted mode, the media stream is not encrypted. Also, if an intervening device (such as a transcoder or gateway) is not enabled for encryption, the media stream is not encrypted.

Disabling and Reenabling Security

The authentication and encryption features between Cisco Unity Connection and Cisco Unified CM can be enabled and disabled by changing the Security Mode for all Cisco Unified CM clusters to Non-Secure, and by changing the applicable settings in the Cisco Unified Communications Manager Administration.

Authentication and encryption can be reenabled by changing the Security Mode to Authenticated or Encrypted.

Note After disabling or re-enabling authentication and encryption, it is not necessary to export the Connection server root certificate and copy it to all Cisco Unified CM servers.

Multiple Clusters Can Have Different Security Mode Settings

When Cisco Unity Connection has multiple Cisco Unified CM phone system integrations, each Cisco Unified CM phone system integration can have different Security Mode settings. For example, one Cisco Unified CM phone system integration can be set to Encrypted, and a second Cisco Unified CM phone system integration can be set to Non-Secure.

Settings for Individual Voice Messaging Ports

For troubleshooting purposes, authentication and encryption for Cisco Unity Connection voice messaging ports can be individually enabled and disabled. At all other times, we recommend that the Security Mode setting for all individual voice messaging ports in a Cisco Unified CM port group be the same.

Packetization

The Real-Time Transport Protocol (RTP) is used to send and receive audio packets over the IP network. Each discrete packet has a fixed-size header, but the packets themselves can vary in size, depending on the size of the audio stream to be transported (which varies by codec) and the packetization setting. This variable size function helps utilize network bandwidth more efficiently. Reducing the number of packets that are created per call sends fewer total bytes over the network.

Packetization is set in the Cisco Unified CM Service Parameters, in the Preferred G711 Millisecond PacketSize and Preferred G729 Millisecond PacketSize parameters. Cisco Unity Connection supports any packet size up to 30ms for G.711 audio, and any packet size up to 60 ms for G.729a audio. The default setting is 20ms for both; there may be latency issues with lower settings.

DSCP is a priority setting on each packet. DSCP helps intermediary routers manage network congestion and lets them know which packets to prioritize ahead of others. Following Cisco AVVID standards, Connection marks the SCCP and SIP packets (call control) with a default DSCP value of 24 (the TOS octet is 0x60), and the RTP packets (audio traffic) with a default DSCP value of 46 (the TOS octet is 0xB8). Thus, the RTP audio packets can be assigned priority over other packets by using the router settings. Note that even though Cisco Unified CM allows you set different DSCP values, when integrated with Connection, the DSCP values set by Connection always take precedence. The marking of both SCCP and SIP packets is configurable in Connection on the System Settings > Advanced > Telephony Configuration page in Cisco Unity Connection Administration.

With each new audio stream (once per call), Cisco Unified CM tells Connection which packet size to use, and Connection sets the DSCP priority for the stream. The entire stream (call) stays at the specified packet size and priority. For example, an audio stream could be broken up into packets of 30ms each. A 30ms G.729a audio stream would be 30 bytes plus the header per packet, and a 30ms G.711 stream would be 240 bytes plus the header per packet. For information on setting Cisco Unified CM Service Parameters, see the Cisco Unified CM documentation at http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html
.

On the Telephony Integrations > Port Group > Port Group Basics > Edit Servers page, you list the Cisco Unified CM servers in different orders:

– For the first port group, the Cisco Unified CM servers are listed in the order specified in the applicable chapter of the
Cisco Unified Communications Manager SCCP Integration Guide for Cisco Unity Connection Release 8.x
.

– For the second port group, the Cisco Unified CM servers are listed in the reverse order.

Cisco Unity Connection supports IPv4, IPv6, or dual-stack (IPv4 and IPv6) addressing with Cisco Unified Communications Manager phone system integrations via SCCP or SIP. When IPv6 is enabled, Connection can obtain an IPv6 address either through router advertisement, through DHCP, or by manually configuring an address either in Cisco Unified Operating System Administration or by using the command-line interface.

For SCCP integrations with Cisco Unified CM, if Connection is configured to listen for incoming IPv4 and IPv6 traffic, you can configure the addressing mode that Connection uses for call control signaling for each port group to use either IPv4 or IPv6. (This mode is also used when connecting to a TFTP server.)

For SIP integrations with Cisco Unified CM, if Connection is configured to listen for incoming IPv4 and IPv6 traffic, you can configure the addressing mode that Connection uses for call control signaling for each port group to use either IPv4 or IPv6. (This mode is also used when connecting to a TFTP server.) In addition, you can configure the addressing mode that Connection uses for media for each port group to use either IPv4 or IPv6.

Note the following considerations when deploying IPv6 for Cisco Unified CM integrations:

The CTL file required for security features (authentication and encryption) between Connection and Cisco Unified CM for SCCP integrations uses IPv4 addressing. Therefore, in order to use authentication and/or encryption with SCCP, you must use either IPv4 or dual-stack addressing.

Some versions of Cisco Adaptive Security Appliance (ASA) do not support application inspection for IPv6 traffic for Unified Communications application servers and endpoints. Cisco recommends not using IPv6 for Unified Communications if you are using a Cisco ASA version that does not provide this support. See the documentation for your version of Cisco ASA to determine whether application inspection is supported in your deployment.

Multiple Cisco Unified Communications Manager Express Version Support

A single Cisco Unity Connection server can support multiple versions of Cisco Unified CM Express. The version of Connection being used must support all versions of Cisco Unified CM Express. See the following documents:

A single, centralized Cisco Unity Connection server can be used by multiple Cisco Unified CM Express routers. This configuration requires that one Cisco Unified CM Express router be on the same LAN as the Connection server, and that this Cisco Unified CM Express router register all Connection voice messaging ports. This Cisco Unified CM Express router (the SIP MWI server) is a proxy server that relays SIP MWI messages between the Connection server and all other Cisco Unified CM Express routers (the SIP MWI clients). Note that Connection voice messaging ports register only with the SIP MWI server (the Cisco Unified CM Express router that is on the same LAN as the Connection server), not with the SIP MWI clients. See Figure 10-9.

A single Cisco Unity Connection server can support multiple versions of Cisco Unified Communications Manager and Cisco Unified Communications Manager Express. The Connection version must support all versions of Cisco Unified CM and/or Cisco Unified CM Express. See the following documents:

When the WAN is down and Connection has foreign exchange office (FXO) or foreign exchange station (FXS) access to a public switched telephone network (PSTN), Connection uses in-band dual tone multifrequency (DTMF) signaling (see Figure 10-11).

Figure 10-11 Cisco Unified Communications Manager Fallback with PSTN

In both configurations, phone message buttons remain active and calls to busy or unanswered numbers are forwarded to Connection. The installer must configure access from the dial peers to the voice-mail system, and establish routing to Connection for busy and unanswered calls and for the message button.

If Connection is accessed over FXO or FXS, you must configure instructions (DTMF patterns) for Connection so it can access the correct voice-mail system mailbox.

When using Cisco Unified SRST with Connection, the integration has the following limitations during a WAN outage:

Call forward to busy greeting
—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Connection, the busy greeting cannot play.

Call forward to internal greeting
—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Connection, the internal greeting cannot play. Because the PSTN provides the calling number of the FXO line, the caller is not identified as a user.

Call transfers
—Because an access code is needed to reach the PSTN, call transfers from Connection to a branch office will fail.

Identified user messaging
—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a user at a branch office leaves a message or forwards a call, the user is not identified. The caller appears as an unidentified caller.

Message waiting indication
—MWIs are not updated on branch office phones, so MWIs will not correctly reflect when new messages arrive or when all messages have been listened to. We recommend resynchronizing MWIs after the WAN link is reestablished.

Message notification
—Because an access code is needed to reach the PSTN, message notifications from Connection to a branch office will fail.

Routing rules
—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call arrives from a branch office to Connection (either a direct or forwarded call), routing rules will fail.

When the Cisco Unified SRST router uses PRI or BRI connections, the caller ID for calls from a branch office to Connection may be the full number (exchange plus extension) provided by the PSTN and therefore may not match the extension of the Connection user. In this case, you can let Connection recognize the caller ID by using alternate extensions.

When using Cisco Unified SRST, Redirected Dialed Number Information Service (RDNIS) must be supported.

Impact of Non-Delivery of RDNIS on Voice Mail Calls Routed by Using AAR

RDNIS must be supported when using Automated Alternate Routing (AAR).

AAR can route calls over the PSTN when the WAN is oversubscribed. However, when calls are rerouted over the PSTN, RDNIS can be affected. Incorrect RDNIS information can affect voice mail calls that are rerouted over the PSTN by AAR when Cisco Unity Connection is remote from its messaging clients. If the RDNIS information is not correct, the caller does not reach the mailbox of the dialed user but instead hears the automated attendant prompt, and might be asked to reenter the extension number of the party the caller wants to reach. This behavior is primarily an issue when the phone carrier is unable to ensure RDNIS across the network. There are numerous reasons why the carrier might not be able to ensure that RDNIS is properly sent. Check with your carrier to determine whether it provides guaranteed RDNIS delivery end-to-end for your circuits. The alternative to using AAR for oversubscribed WANs is simply to let callers hear reorder tone in an oversubscribed condition.

Cisco Unity Connection supports a topology with centralized call processing and distributed messaging, in which your Connection server is located at a remote site or branch office and registered with Cisco Unified CM at a central site.

When the WAN link fails, the phones fall back to the Cisco Unified CM Express-as-SRST device. Connection can also fall back to the Cisco Unified CM Express-as-SRST device, which lets users at the remote site access their voice messages and see message waiting indicators (MWIs) during a WAN outage. Note that MWIs must be resynchronized from the Connection server whenever a failover happens from Cisco Unified CM to Cisco Unified CM Express-as-SRST or vice versa.

Survivable Remote Site Voicemail

Survivable Remote Site Voicemail (SRSV) provides backup voicemail service in a centralized messaging and centralized call processing deployment. For example, Figure 10-12 shows SRSV using Cisco Unity Express in a branch location to provide backup voicemail service for Cisco Unity Connection in a headquarters location when the connection between the sites is unavailable. During normal operations, Cisco Unified Messaging Gateway in the headquarters location retrieves configurations (for example, SRST phones, users, and mailbox information) from Cisco Unified CM and Cisco Unity Connection to provision and update the mailboxes in Cisco Unity Express SRSV, based on a configured schedule. Cisco Unity Express SRSV is active only when SRST is activated, and it remains idle otherwise. When the network connection between sites is restored, Cisco Unity Express SRSV uploads all messages (new, saved, deleted, and so on) to Cisco Unity Connection.

Integrating by Using SIP

SIP (Session Initiation Protocol) is the Internet Engineering Task Force standard for multimedia calls over IP. SIP is a peer-to-peer, ASCII-based protocol that uses requests and responses to establish, maintain, and terminate calls (or sessions) between two or more end points. See
Table 10-5.

Table 10-5 SIP Network Components

Component

Description

SIP proxy server

An intermediate device that receives SIP requests from a client and then forwards the requests on behalf of the client. Proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.

Redirect server

Provides information to the client about the next hop or hops that a message should take. The client then contacts the next hop server or user-agent server directly.

Registrar server

Processes requests from user agent clients for registration of their current location. Registrar servers are often installed on the redirect or proxy server.

Phones

Acts as either a server or client. Softphones (PCs that have phone capabilities installed) and Cisco SIP IP phones can initiate SIP requests and respond to requests.

Gateways

Provide call control. Gateways provide many services; the most common is a translation function between SIP call endpoints and other terminal types. This function includes translation between transmission formats and between communications procedures. In addition, the gateway translates between audio and video codecs, and performs call setup and clearing on both the LAN side and the switched-circuit network side.

SIP uses a request/response method to establish communications between various components in the network and to ultimately establish a conference (call or session) between two or more endpoints. A single call may involve several clients and servers.

Users in a SIP network are identified by:

A unique phone or extension number.

A unique SIP address, which is similar to an email address and uses the format sip:<userID>@<domain>. The user ID can be either a user name or an E.164 address.

When a user initiates a call, a SIP request typically goes to a SIP server (either a proxy server or a redirect server). The request includes the caller address (From) and the address of the called party (To).

SIP messages are in text format using ISO 10646 in UTF-8 encoding (like HTML). In addition to the address information, a SIP message contains a start-line specifying the method and the protocol, a number of header fields specifying call properties and service information, and an optional message body which can contain a session description.

Note Cisco Unity Connection extracts Caller Id from Remote Party Id field and FROM field in SIP Invite. In addition, when the Remote Party Id option is unchecked on CUCM SIP trunk and the FROM field is set to Anonymous in a SIP header, Connection treats the caller as unknown.

Description of PIMG Integrations

The PIMG integration uses one or more PIMG units between the circuit-switched phone systems and IP network. On the circuit-switched phone system side, there are both digital (feature-set) and analog interfaces; the interface used depends on the phone system to which Cisco Unity Connection is connected. On the IP side, there is a SIP interface, which is how Connection communicates with the PIMG units. To Connection, the integration is essentially a SIP integration. Connection communicates with the PIMG units over the IP network by using SIP and RTP protocols. The PIMG units communicate with the circuit-switched phone system over the phone network using phone system-specific protocols (digital, analog, or serial).

Firmware Updates

Note that when receiving shipment of PIMG or TIMG units, it may be necessary to update the firmware on the units. The PIMG/TIMG Administration interface provides a simple method to update the firmware files. Firmware updates are available at http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=278875240
(note that you must sign in to www.cisco.com to access the URL). For details, see the applicable integration guide.

Serial Integrations

Cisco Unity Connection supports the following serial protocols:

SMDI

MCI

MD-110

The serial port on PIMG/TIMG units was originally designed as a management port rather than as a standard RS-232 serial port. Consequently, a custom serial cable (which is available from Cisco) is necessary for the data link between the phone system and the master PIMG/TIMG unit.

Increasing Port Capacity

PIMG units have eight ports. To increase system port capacity, multiple PIMG units can be stacked. For example, if 32 ports are needed, four PIMG units can be stacked.

TIMG units, which integrate with circuit-switched phone systems that support T1-CAS, have 24 T1 ports per span in a single rack-optimized unit. Single-span, dual-span, and quad-span TIMG units are available.

Multiple Integration Support/Branch Office Consolidation

Revised August 6, 2013

PIMG/TIMG units can be separated by a WAN to support circuit-switched phone systems at remote branch office sites. For example, Cisco Unity Connection could be placed at a centralized headquarters and support circuit-switched phone systems both at the headquarters and at the branch office sites.

As an example, assuming there are four phone systems from four different manufacturers (for example, Nortel, Avaya, NEC, and Siemens), four different phone system integrations could be created on the Connection server to support the four phone systems. A standalone Connection server supports up to 250 ports that will connect to the four phone systems. For example:

At the Seattle site, 19 PIMG units can be stacked to support 150 ports.

At the New York site, four PIMG units can be stacked to support 30 ports.

At the Tokyo site, seven PIMG unit can be used to support 50 ports.

At the Dallas site, three PIMG unit can be used to support 20 ports.

Note that even though the PIMG units come with eight ports, fewer than eight ports can be used on each unit.

If PIMG units will be separated by a WAN to support remote phone systems, correct audio codec selection, bandwidth capacity planning, and QOS planning are required. Both the G.729a and G.711 audio codecs are supported by PIMG units and by Connection. Because PIMG units are Dialogic devices rather than Cisco devices, the use of location-based CAC is not applicable. The following network and bandwidth requirements are required when placing the PIMG across a WAN:

When PIMG units are separated by a WAN, prioritize your call control and media traffic through proper QOS traffic, marking for voice traffic originating on the PIMG units. Set the Call Control QOS Byte and RTP QOS Byte on PIMG units to the following values:

In the Call Control QOS Byte field, enter 104.

In the RTP QOS Byte field, enter 184.

Note that the Call Control QOS Byte and RTP QOS Byte fields on PIMG units define a decimal value that represents QOS bit flags. These values can be interpreted as either IPv4 TOS or Differentiated Services Codepoint (DSCP). For more details, see the
Dialogic 1000 and 2000 Media Gateway Series User’s Guide
, provided by Dialogic.

There must be an adequate number of voice messaging ports on the Connection server to connect to the phone systems. This number of ports must not exceed the number of ports that are enabled by the Connection license files.

Connection is installed on a separate server from Cisco Unified CM. Multiple integrations are not supported when Connection is installed as Cisco Unified Communications Manager Business Edition (CMBE)—on the same server with Cisco Unified CM.

Optional Integration Features

Alternate Extensions

In addition to the primary extension for each user, you can set up alternate extensions. Alternate extensions can be used for various reasons, such as handling multiple line appearances on user phones. Alternate extensions can also make calling Cisco Unity Connection from an alternate device—such as a mobile phone, a home phone, or a phone at another work site—more convenient.

When you specify the phone number for an alternative extension, Connection handles all calls from that number in the same way that it handles calls from a primary extension (assuming that ANI or caller ID is passed along to Connection from the phone system). This means that Connection associates the alternate phone number with the user account, and when a call comes from that number, Connection prompts the user to enter a password and sign in.

Alternate MWIs

You can set up Cisco Unity Connection to activate alternate MWIs when you want a new message for a user to activate the MWIs at up to 10 extensions. For example, a message left at extension 1001 can activate the MWIs on extensions 1001 and 1002.

Connection uses MWIs to alert the user to new voice messages. MWIs are not used to indicate new email, fax, or return receipt messages.

Centralized Voice Messaging

Cisco Unity Connection supports centralized voice messaging through the phone system, which supports various inter-phone system networking protocols including proprietary protocols such as Avaya DCS, Nortel MCDN, or Siemens CorNet, and standards-based protocols such as QSIG or DPNSS. Note that centralized voice messaging is a function of the phone system and its inter-phone system networking, not voice mail. Connection will support centralized voice messaging as long as the phone system and its inter-phone system networking are properly configured.

When discussing phone systems involved in centralized voice messaging, there are essentially two types:

Message Center PINX
—The phone system hosts the voice messaging system (the phone system is directly connected to the voice messaging system).

User PINX
—The phone system is remote from the voice messaging system (the phone system is not directly connected to the voice messaging system).

Centralized voice messaging provides voice messaging services to all users in a networked phone system environment. Connection can be hosted on a message center PINX and provide voice messaging services to all users in an enterprise assuming the message center PINX and all user PINX phone systems are properly networked.

For a centralized voice messaging configuration to exist, a suitable inter-phone system networking protocol must exist to deliver a minimum level of feature support, such as:

Message waiting indication (MWI).

Transfer, which ensures that the correct calling/called party ID is delivered to the voice messaging system.

Divert, which ensures that the correct calling/called party ID is delivered to the voice messaging system.

Other features may be required depending on how the voice messaging system is to be used. For example, if it is also serving as an automated attendant, path-replacement is needed as this feature prevents calls from hair-pinning.

Not all phone systems can serve as a message center PINX. In this case, customers may wish to consider relocating Connection to Cisco Unified Communications Manager and have Cisco Unified CM act as the message center PINX with the circuit-switched phone system now acting as the user PINX.

For information on configuring Connection in a centralized voice messaging environment to be hosted on Cisco Unified CM serving as the message center PINX, see the following:

Note that if customers are deploying centralized voice messaging with Connection and a circuit-switched phone system, it is up to the customer to determine whether the circuit-switched phone system can serve as a message center PINX on which Connection can be hosted. If so, the customer should also confirm that there is support for the desired features, for example, MWIs, transfer, divert, and path-replacement.

Inter-cluster trunks between Cisco Unified CM clusters can be QSIG-enabled by using the Annex M.1 feature, which allows Connection to integrate with a single Cisco Unified CM cluster. Ports in the cluster with which Connection is integrated can be dedicated to turning MWIs on and off for phones in other clusters.

Integrating Cisco Unity Connection with a QSIG-Enabled Phone System by Using Cisco ISR Voice Gateways

Cisco Unity Connection supports an integration with a QSIG-enabled phone system through a Cisco ISR voice gateway. See Figure 10-13.

Figure 10-13 Connections Between the Phone System and Cisco Unity Connection